US8233485B2 - Network interoperability - Google Patents
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- US8233485B2 US8233485B2 US11/892,033 US89203307A US8233485B2 US 8233485 B2 US8233485 B2 US 8233485B2 US 89203307 A US89203307 A US 89203307A US 8233485 B2 US8233485 B2 US 8233485B2
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/102—Gateways
- H04L65/1033—Signalling gateways
- H04L65/104—Signalling gateways in the network
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/102—Gateways
- H04L65/1023—Media gateways
- H04L65/103—Media gateways in the network
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
- H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04Q—SELECTING
- H04Q3/00—Selecting arrangements
- H04Q3/0016—Arrangements providing connection between exchanges
- H04Q3/0029—Provisions for intelligent networking
- H04Q3/0045—Provisions for intelligent networking involving hybrid, i.e. a mixture of public and private, or multi-vendor systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/1016—IP multimedia subsystem [IMS]
Definitions
- the present invention relates to interoperability between communication networks. More specifically, it relates to a situation in which application servers of an internet protocol multimedia subsystem (IMS) provide services to terminals of a time division multiplexing (TDM) intelligent network (IN).
- IMS internet protocol multimedia subsystem
- TDM time division multiplexing intelligent network
- Network service providers are facing a major migration challenge, as the telephony networks currently deployed using TDM digital switches need to be redesigned to use voice over internet protocol (VoIP).
- Emerging standards such as the European Telecommunication Standards Institute (ETSI) 3 rd Generation Partnership (3GPP) IMS also change the way network services may be designed.
- ETSI European Telecommunication Standards Institute
- 3GPP 3 rd Generation Partnership
- a call from a user located in a city A to another user also in the city A may need to be processed by a centralized switch located in a city B. If the distance between these two cities is large, then the voice path is long enough to introduce perceptible delays, echoes, and require additional and expensive echo cancellers.
- the widely used signaling system #7 does not solve the problem even if the transport networks are separate for signaling and voice bearers.
- the transport network of the signaling and voice are indeed separate, the SS7 integrated services digital network user part (ISUP) signaling and voice bearers are still processed by each switch.
- ISUP integrated services digital network user part
- the technology called intelligent network (IN) can be used in order to partially solve this problem.
- This technology uses an abstract call model, and lets an external application interact with a remote TDM switch using control primitives acting on the abstract call model.
- the intelligent network application part (INAP) protocol standardized by the International Telecommunications Union (ITU) (ITU-T SG 11, recommendation Q1224 approved in 1997) is the most widely used protocol enabling an external application to interact with remote TDM switches using a call processing logic that can be modeled according to the state machine defined by ITU Q1224.
- the ITU has also defined IN capability set (CS) call models (ITU-T SG 11 Q1244 defines CS4), which let INAP applications control voice over IP (VoIP) calls.
- CS IN capability set
- FIG. 1 represents a simplified block diagram of a prior art TDM IN 100 solution.
- SSP service switching point
- SSF service switching function
- SCP service control point
- SCF service control function
- the distance between A, B and C can be hundreds or even thousands of kilometers creating a potential tromboning problem, and possibly inducing echo. For this reason the SSP 103 is often collocated with local telephone exchanges requiring many points of presence for such SSP function and inducing substantial costs.
- the SSP 103 acts as a trigger point for further services to be invoked during a call.
- the SSP 103 may invoke a query to the SCP 105 to wait for instructions on how to proceed.
- the SCP 105 contains service logic which implements the behavior described by the operator. This enables service providers to deploy the application logic centrally, while only SSPs are distributed.
- the signaling protocol used between the SSP 103 and the SCP 105 is INAP and is based on an abstract call model which represents the capabilities of the SSP 103 .
- the SSP and SCP functions can also be collocated, leading to the so-called “service node” deployment model. In this case there is no longer a requirement for standardized call model. However, this induces even higher costs as the SSP function needs to be distributed because of tromboning issues and therefore, in this case also the application needs to be deployed over multiple Points of Presence (POPs).
- POPs Points of Presence
- VoIP technology separates the media transport plane and the call & media control plane.
- a VoIP service switch can interact with media streams without processing the IP packets containing the signaling at all.
- the SSP function can now be centralized and because it can be centralized it can also be collocated with the application logic itself (SCP). With both SSP and SCP collocated, there is no longer a need for a standardized call model supporting communication between them.
- ETSI 3 rd Generation Partnership Project (3GPP) defined internet protocol multimedia subsystem (IMS) which is a new telecommunications service architecture optimized for VoIP technology.
- FIG. 2 shows a simplified block diagram of an IMS network 200 .
- An IP network 201 is connected to a call session control function (CSCF) block 203 .
- the CSCF block 203 further comprises serving CSCF (S-CSCF) 205 , interrogating CSCF (I-CSCF) 207 and proxy CSCF (P-CSCF) 209 .
- the CSCF block 203 is connected to application servers (ASs) 213 .
- the IMS calls are processed by one or several ASs under control of the S-CSCF 205 .
- the CSCF block 203 is further connected to home subscriber server (HSS) 211 . It contains the subscriber-related information, performs authentication and authorization of the user, and can provide information of the physical location of the user.
- HSS home subscriber server
- the CSCF block 203 is further connected to a media resource function (MRF) 215 and a gateway 217 , which interfaces with a circuit switched public switched telephone network (PSTN) or public land mobile network (PLMN) 219 .
- the MRF 215 provides a source of media in the home network and it is, for instance used for playing announcements, real-time transcoding of multimedia data, text-to-speech conversation and speech recognition.
- VoIP Voice over IP Services
- TDM networks required a standardized call model in order to allow applications to efficiently interact with the communications network
- ASs 213 have access to the call signaling directly, and therefore no call model is required. This gives much more flexibility to application designers, reduces the complexity of the network, mainly by reducing the level of interactions between components, and makes it possible to introduce applications not previously allowed by the call models standardized for TDM networks.
- the IMS network 200 allows support for applications in a roaming environment for cellular phones.
- TDM INs 100 most services, e.g. a short numbering plan for a corporation, are lost when roaming outside of the home network, because routing outgoing calls to the home network would create a tromboning problem, and because INAP links are not so frequently allowed among roaming networks.
- an IMS network 200 on the contrary, the signaling of all calls placed by an end user in a roaming situation is routed to the home network, and therefore most services working in the home environment will work in a roaming situation.
- TDM networks such as mobile GSM networks and IMS networks 200 will have to coexist. It is plausible for TDM service providers to start developing all new applications in an IMS mode, using IMS ASs 213 , and stop developing IN applications in current TDM networks. However, most IMS devices will have to work in dual mode initially, and therefore the same application logic needs to be available while working in IMS mode or working in TDM mode.
- One object of the invention is to limit the above identified deficiencies. More specifically, a method for connecting the IMS network 200 to the IN 100 by use of ASs 213 has been invented.
- a method for connecting at least one AS of an IMS network to an IN through an interface unit wherein the IMS comprises a CSCF unit connected to the at least one AS arranged for processing call signaling, wherein the method comprises the following steps with respect to a call:
- the invention in accordance with an embodiment of the invention has the advantage that it allows a smooth integration of the IN 100 and the IMS network 200 .
- the invention thus makes it possible to the IMS ASs 213 to control the IN 100 endpoints. Furthermore, there is no need to design the applications separately to the INs 100 and to the IMS networks 200 , nor is there a need to implement the IN call model in IMS ASs 213 and develop the applications in IN mode only.
- the invention further allows, in accordance with an embodiment, to program telecommunication applications using only the SIP/IMS service control (ISC) interface.
- the interface can take control of heterogeneous IMS and TDM/IN endpoints, and the applications do not need to handle any special case for the TDM to IP or IP to TDM calls.
- the invention also relates to a corresponding system, an interface unit and a computer program product.
- FIG. 1 is a simplified block diagram of a prior art TDM intelligent network 100 ;
- FIG. 2 is a simplified block diagram of a prior art IMS network 200 ;
- FIG. 3 is a simplified block diagram of an architecture for integrating the IN 100 and the IMS network 200 in accordance with an embodiment of the present invention.
- FIG. 4 is a flow chart depicting a method for integrating the IN 100 and the IMS network 200 in accordance with an embodiment of the invention.
- the originating side is the side and corresponding network elements from where the call originates as opposed to terminating side which is the side and corresponding network elements where the call terminates.
- the TDM IN 100 operates in accordance with global system for mobile communications customized applications for mobile network enhanced logic (GSM CAMEL) standard, but the IN 100 could equally use any other TDM communication standard.
- GSM CAMEL mobile network enhanced logic
- FIG. 3 represents how the IN 100 shown in FIG. 1 and the IMS network of FIG. 2 can be connected to each other via an interface unit (IU) 301 .
- the session initialization protocol (SIP) used by IMS ASs 213 has been extended by the ETSI 3GPP in order to introduce facilities not present in the original Internet Engineering Task Force (IETF) SIP protocol, in particular to facilitate the chaining of applications and the control of application sequencing by the S-CSCF 205 .
- the S-CSCF 205 and the AS 213 are thus equipped with IMS service control (ISC) interfaces to communicate with each other and to provide the extended SIP facilities.
- ISC IMS service control
- the IU 301 uses a first access protocol, in this example ISC/SIP to communicate with the S-CSCF 205 and a second access protocol, in this case INAP protocol, to communicate with the IN 100 .
- a first access protocol in this example ISC/SIP to communicate with the S-CSCF 205
- a second access protocol in this case INAP protocol
- the IU 301 of FIG. 3 is interfacing with an external S-CSCF 205 .
- an external S-CSCF 205 it is also possible to embed a dedicated S-CSCF function into the IU 301 . In the latter case the IU 301 behaves like a standard IMS S-CSCF 205 .
- the S-CSCF 205 is responsible for sending the IMS call over the ISC interface to each of the relevant ASs 213 .
- ASs 213 are instructed by the S-CSCF 205 to return the call signaling, after appropriate processing, to the S-CSCF 205 , which can then send the call signaling to the next AS 213 .
- This property is used to enable access to many applications designed on top of an IMS AS 213 from a TDM IN 100 . This is achieved by emulating IMS entities on top of an IN SCP 105 .
- the IU 301 when invoked by the IN 100 is arranged to give instructions regarding the processing of a call, to simulate a 3GPP SIP call to the target AS(s) 213 relayed by a P-CSCF 209 if the call is from a TDM terminal or I-CSCF 207 if the call is directed to a TDM terminal.
- the calls of subscribers get specific processing depending on whether the call originates from the IN 100 or from the IMS network 200 .
- the call control is directed to the IU 301 which further relays the call to the S-CSCF 205 .
- the call will get back to the S-CSCF 205 after processing by each AS 213 , and the S-CSCF 205 routes this call to the IU 301 as last originating side AS 213 .
- This enables the IU 301 to analyze the modifications affecting the call signaling after processing by the ASs 213 , and convert these modifications as appropriate for the I N call model, if possible and necessary.
- the call For incoming calls from the IMS network 200 , the call will reach the IU 301 , because the IU 301 registers the SIP address of record (AoR, the public identity of the endpoint) of the endpoints it controls onto the S-CSCF 205 .
- AoR the public identity of the endpoint
- the IU 301 sends TDM calls received on the terminating side, i.e. to the GSM user, to the IMS network 200 . This is achieved by the IU 301 emulating the I-CSCF 207 sending a call from another network or S-CSCF 205 to the S-CSCF 205 of the target user.
- IMS applications can be converted in an optimal way to proper INAP primitives.
- an IMS service implementing a short numbering plan for an enterprise modifies only calling and called numbers, or generates announcements.
- the IU 301 can detect whether a call is GSM to GSM and in this case sends the customized applications for mobile network enhanced logic (CAMEL), an IN protocol with extensions for GSM networks, instruction to the GSM mobile switching centre (MSC), a TDM switch with an IN call model, to route the call after modification of the calling or called parties.
- CAMEL mobile network enhanced logic
- MSC GSM mobile switching centre
- TDM switch with an IN call model
- a user of an application using the IU 301 will need to have a CAMEL trigger defined for all relevant incoming (DP 12 ) and outgoing (DP 2 ) calls. This is defined in the GSM home location register (HLR) as part of the profile of that user.
- HLR GSM home location register
- the IN switches also need to be configured to send an INAP initial detection point to the IU 301 for any outgoing or incoming call. The way to configure this depends on the TDM network vendor, but usually involves a “service mark”, e.g. specific configuration information in the access TDM switch, linked to the physical access link to the user.
- the IU 301 may decide to execute the application logic in the IN network, or later when the call is processed by the VoIP network. If the AS 213 application logic generates an announcement, then the IU 301 instructs the MSC to connect the call to the VoIP announcement server through a VoIP gateway.
- the IU 301 is not invoked for VoIP to VoIP calls, as the application logic used will be that of the IMS AS 213 , and does not require any adaptation.
- the IU 301 identifies TDM to TDM calls, and emulates the IMS AS application logic using INAP commands to the TDM IN switches of the calling and called party users.
- the IU 301 needs to be made aware of any outgoing or incoming calls from terminals having the phone numbers that must be controlled by the IMS network 200 . This is achieved by registering the IU 301 logic as INAP trigger detection point (TDP-R).
- TDP-R INAP trigger detection point
- Calls are routed through a voice over internet gateway over the IMS 200 at least in one of the following situations: TDM to IP calls, IP to TDM calls or when call handling capabilities of the AS exceed the capabilities of a SSP 103 of the IN 100 .
- the call routing is done by the IN 100 instructing an SSP 103 of the IN 100 to route the call through a voice over internet gateway over the IMS 200 .
- the destination is reachable through the TDM IN 100 and the call is received from the IN 100 , which operates in accordance with a GSM standard.
- the IU 301 receives an Initial DP service request, i.e. an incoming call from the originating IN SSP 103 of an incoming call from the IN 100 . Then at step 402 by emulating the operation of the P-CSCF 209 , or access gateway control function (AGCF) in the TISPAN IMS variant, the IU 301 sends a SIP INVITE message to the S-CSCF 205 of this user (possibly through an I-CSCF 207 if the service provider has so configured its IMS network 200 ).
- an Initial DP service request i.e. an incoming call from the originating IN SSP 103 of an incoming call from the IN 100 . Then at step 402 by emulating the operation of the P-CSCF 209 , or access gateway control function (AGCF) in the TISPAN IMS variant, the IU 301 sends a SIP INVITE message to the S-CSCF 205 of this user (possibly through an I-CSCF
- the S-CSCF 205 then executes the service triggers defined for this user, by propagating (step 403 ) the SIP INVITE message to a sequence of originating ASs 213 which may relay and modify (step 404 ) any parameter of the SIP INVITE message, or even absorb it.
- the last AS 213 sends the processed SIP INVITE message to the S-CSCF 205 and S-CSCF 205 forwards (step 406 ) the processed SIP INVITE message to the IU 301 .
- This is achieved by defining the IU 301 as the last originating AS 213 for this user's identity in the user HSS 211 profile, in this case originating and terminating sides are handled separately, but it is also possible to wait until the SIP call returns to the terminating side IU 301 , in which case the IU 301 will execute in one step both originating and terminating services. This is called here as a variation “SIMPLE”.
- the IU 301 accesses the processed SIP INVITE message parameters as modified by the sequence of originating ASs 213 invoked by the S-CSCF 205 .
- the originating side IU 301 when it receives the SIP INVITE message processed by the sequence of originating side ASs 213 , analyses (step 408 ) the new destination number or uniform resource identifier (URI) as modified by the ASs 213 .
- the ASs 213 may have expanded a short number, e.g. 1234, into a public format, such as +33 671201234.
- the IU 301 determines (step 409 ) that the destination number, at this stage, is reachable in TDM mode by the SSP 103 , it instructs the SSP 103 to propagate (step 410 ) the call over TDM IN 100 using the appropriate CAMEL commands.
- the reachability may be determined by multiple means, e.g. by using number pattern analysis, i.e. number belongs to a number block which is part of the IN 100 , local number portability query, carrier for this number corresponds to the IN 100 , or HLR query for GSM networks. Several of these reachability determination methods can be used simultaneously.
- the destination user may have set its profile to redirect the call to another destination. If the new destination is TDM or if the call has not been redirected, then the call flow is optimized, end-to-end, over TDM. However, if the new destination is IP, then a VoIP gateway will be invoked on the terminating side. This can be achieved by instructing the IN SSP to route the call to a VoIP gateway acting as an Intelligent Peripheral. The MRF 215 could act for this purpose as the VoIP gateway. This method effectively gives precedence to the TDM path over the IP path by the originating IU 301 .
- the destination is an endpoint also controlled by the IMS service logic
- its IU 301 will be invoked by a CAMEL Initial DP command from the SSP 103 (running on the originating side in the CAMEL model).
- the IU 301 managing the destination endpoint sends a SIP INVITE message to the terminating side S-CSCF 205 of the corresponding user identity, and the terminating side S-CSCF 205 invokes the terminating side ASs 213 as configured in the HSS profile of this user's identity.
- the terminating side IU 301 then receives an SIP INVITE message processed by the sequence of terminating side ASs 213 , because the IU 301 registered the corresponding address of record.
- the terminating IU 301 communicates modified call parameters to the originating IU 301 at this stage, triggering the routing process described earlier.
- the terminating side IU 301 instructs the SSP 103 to connect the call if the modified destination is reachable by this SSP 103 .
- Any of the originating side ASs 213 may terminate the call locally, e.g. to an announcement specifying that the call has been blocked.
- the IU 301 does not receive the SIP INVITE message of step 406 .
- the originating side IU 301 receives a session description protocol (SDP) answer, and connects the GSM terminal to the announcement through a VoIP gateway.
- SDP session description protocol
- any of the terminating side ASs 213 terminates the call locally, e.g. to an announcement specifying that the call has been blocked.
- the originating side IU 301 determines at step 409 that the destination number, at this level, is not reachable in TDM mode by the SSP 103 , it acts as a non-concerned AS 213 by returning (step 411 ) the SIP INVITE message to the S-CSCF 205 unmodified with an SIP contact header allowing the S-CSCF 205 to by-pass this AS 213 for any subsequent message.
- the terminating side S-CSCF 205 redirects (step 412 ) the call to another destination. If the destination is in IP network, as is the case for a normal call not redirected by the destination, then the TDM to IP call flow is optimized, and the VoIP gateway used is that controlled by the originating IU 301 and the MRF 215 acts as a VoIP gateway.
- the TDM shortcut will be established by the originating side IU 301 if it runs on the same IU 301 as the terminating side IU 301 .
- An incoming call to an IN 100 controlled endpoint may be coming from 3 types of callers:
- mapping to IN 100 can be performed provided that the application logic complies with the following limitations:
- the invention relates to a corresponding system that comprises at least the IU 301 that is arranged to perform any of the method steps in accordance with the embodiments of the invention.
- the system may further comprise at least some elements of the IN 100 and the IMS network 200 .
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- Computer Networks & Wireless Communication (AREA)
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- Data Exchanges In Wide-Area Networks (AREA)
Abstract
Description
-
- the interface unit directing call signaling to the at least one AS through the CSCF unit;
- the at least one AS processing the call signaling and sending the processed call signaling through the CSCF unit back to the interface unit; and
- based on the information received in the call signaling processed by the at least one AS, the interface unit directing the incoming call to a destination.
-
- A caller using an endpoint managed by the
IN 100. This case has been explained above. - A caller using an endpoint controlled by the IMS network. In this case, the terminating side services of the called endpoint are executed in the
IMS network 200 before reaching theIU 301. In this case theIU 301 simply routes the call to a gateway MSC of a GSM network, and recognizing this, the gateway MSC simply instructs theSSP 103 to continue normal call handling, thus terminating the INAP control relationship. - A caller using an endpoint managed by a third party network. In this case the call arrives at a regular gateway MSC, and an INAP initial DP is triggered to the
IU 301. Terminating side services have not yet been performed, and theIU 301 needs to emulate an incoming call to theIMS network 200. TheIU 301 then sends an SIP INVITE message to the S-CSCF 205 as described above and rest of the call processing procedure goes according to steps 406-412.
- A caller using an endpoint managed by the
-
- if a first media port presented to the caller is a TDM port, i.e. SDP information corresponding to an IN controlled endpoint, then the ISC application server does not initiate a new SDP offer for the rest of this dialogue, the initial offer/answer SDP exchange is final.
- it is acceptable for the application, if the first media port presented to the caller is an IP port, that any subsequent SDP offer with a TDM port will not be optimized, i.e. the media path between the TDM user devices will be routed via the IMS. Tromboning will occur through IP for the rest of the dialog, even though at some moments during the dialog, the source and destination of media streams are both in the TDM network.
Claims (10)
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EP1858218B1 (en) * | 2006-05-17 | 2011-09-14 | Deutsche Telekom AG | Method and entities for providing call enrichment of voice calls and semantic combination of several service sessions to a virtual combined service session |
US8595642B1 (en) | 2007-10-04 | 2013-11-26 | Great Northern Research, LLC | Multiple shell multi faceted graphical user interface |
US8165886B1 (en) | 2007-10-04 | 2012-04-24 | Great Northern Research LLC | Speech interface system and method for control and interaction with applications on a computing system |
US9843650B2 (en) * | 2009-09-03 | 2017-12-12 | Avaya Inc. | Intelligent module sequencing |
EP2337298A1 (en) * | 2009-11-23 | 2011-06-22 | Alcatel Lucent | R-IM-SSF gateway and method for translating IN protocol messages into SIP messages and vice versa |
EP2326058A1 (en) * | 2009-11-23 | 2011-05-25 | Alcatel Lucent | Method and device for translating IN protocol messages into SIP messages and vice versa |
US20130219070A1 (en) * | 2012-02-16 | 2013-08-22 | Research In Motion Limited | Resolving device specific identifiers to a user identifier to initiate a dialog establishment with devices of a user |
US9602556B1 (en) | 2013-03-15 | 2017-03-21 | CSC Holdings, LLC | PacketCable controller for voice over IP network |
US20210064984A1 (en) * | 2019-08-29 | 2021-03-04 | Sap Se | Engagement prediction using machine learning in digital workplace |
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Also Published As
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EP1890496A1 (en) | 2008-02-20 |
ATE513418T1 (en) | 2011-07-15 |
US20080056242A1 (en) | 2008-03-06 |
EP1890496B1 (en) | 2011-06-15 |
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